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Grandstream Networks, Inc. Analog IP Gateway GXW-410xv 4 or 8 FXO Ports w/Video Surveillance GXW-410xv User’s Manual www.grandstream.com Firmware Version 1.0.0.36 [email protected]

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Grandstream Networks, Inc. Analog IP Gateway GXW-410xv 4 or 8 FXO Ports w/Video Surveillance

GXW-410xv User’s Manual www.grandstream.comFirmware Version 1.0.0.36 [email protected]

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TABLE OF CONTENTS GXW-410XV USER MANUAL

WELCOME....................................................................................................................................................4 PACKAGING ................................................................................................................................................5

SAFETY COMPLIANCES ..............................................................................................................................................5 WARRANTY ...............................................................................................................................................................5 CONNECTING THE GXW-410XV................................................................................................................................5

APPLICATION DESCRIPTION ....................................................................................................................7 SIMPLE CONFIGURATION: MEDIA GATEWAY TO ACCESS PSTN NETWORKS ............................................................7 EXTENSIVE CONFIGURATION: MEDIA GATEWAY CONFIGURATION FOR MULTIPLE USERS.......................................8 OFF-HOOK AUTO DIAL ..............................................................................................................................................8 SAMPLE CONFIGURATIONS - IP PBX PEERS WITH GXW-410XV ...............................................................................8

FEATURES...................................................................................................................................................9 SOFTWARE FEATURES OVERVIEW .............................................................................................................................9 HARDWARE SPECIFICATION.....................................................................................................................................11

CONFIGURATION GUIDE .........................................................................................................................11 CONFIGURATION WITH WEB BROWSER ...................................................................................................................11

Accessing the Web Configuration Menu ............................................................................................................11 End User Configuration.....................................................................................................................................12

ADVANCED USER SETTINGS ....................................................................................................................................13 Advanced User Configuration............................................................................................................................13 Saving the Configuration Changes ....................................................................................................................20 Rebooting from Remote......................................................................................................................................20

VIDEO SURVEILLANCE............................................................................................................................21 VIDEO SURVEILLANCE PROCEDURES.......................................................................................................................21

FIRMWARE UPGRADE .............................................................................................................................22 UPGRADE THROUGH HTTP......................................................................................................................................22 UPGRADE THROUGH TFTP ......................................................................................................................................23

No Local TFTP Server .......................................................................................................................................23 RESTORE FACTORY DEFAULT SETTING..............................................................................................24

FACTORY RESET.................................................................................................................................................24 EXAMPLES OF GXW-410XV CONFIGURATIONS...................................................................................25 GLOSSARY OF TERMS ............................................................................................................................27

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TABLE OF FIGURES GXW-410XV USER MANUAL

FIGURE 1: DIAGRAM OF GXW-410XV BACK PANEL ..........................................................................................5 FIGURE 2: DIAGRAM OF GXW-410XV DISPLAY PANEL......................................................................................6 FIGURE 3: FUNCTIONAL DIAGRAM OF IP-PBX & GXW-410XV ..........................................................................7 FIGURE 4: SCREEN-SHOT OF GXW-410XV LOG-IN SCREEN............................................................................12 FIGURE 5: SCREENSHOT OF ADVANCED USER CONFIGURATION......................................................................13 FIGURE 6: SCREEN-SHOT OF GXW-410XV CHANNEL DIALING ........................................................................16 FIGURE 7: SCREEN-SHOT OF SAVE CONFIGURATION ......................................................................................20 FIGURE 8: SCREEN-SHOT OF REBOOTING ......................................................................................................20 FIGURE 9: SCREEN-SHOT OF VIDEO SURVEILLANCE*......................................................................................22 APPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER........................................................25 APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIO ............................................................25 APPLICATION THREE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER AND VIDEO SURVEILLANCE............26 APPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATION.................................26

TABLE OF TABLES

GXW-410XV USER MANUAL TABLE 1: DEFINITIONS OF THE GXW CONNECTORS.........................................................................................5 TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL ......................................................................................6 TABLE 3: GXW-410XV SOFTWARE FEATURES ..............................................................................................10 TABLE 4: HARDWARE SPECIFICATION OF GXW-410XV ...................................................................................11 TABLE 6: WEB LOG-IN DEFINITIONS ...............................................................................................................12 TABLE 7: STATUS PAGE DEFINITIONS ............................................................................................................13 TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONS ..............................................................................14 TABLE 9: FXO LINES CONFIGURATION DEFINITIONS .......................................................................................16 TABLE 10: CHANNELS PAGE DEFINITIONS ......................................................................................................17 TABLE 11: GRANDSTREAM CONFIGURATION PAGE DEFINITIONS......................................................................18

GUI INTERFACES

GXW-410XV USER MANUAL http://www.grandstream.com/GUI/GUI_GXW410x.rar

1. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE 2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE 3. SCREENSHOT OF CHANNELS CONFIGURATION PAGE 4. SCREENSHOT OF FXO LINES CONFIGURATION PAGE 5. SCREENSHOT OF PROFILE 1 CONFIGURATION PAGE 6. SCREENSHOT OF STATUS CONFIGURATION PAGE

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WELCOME Thank you for purchasing the Grandstream GXW–410xv IP Analog FXO Gateway. The GXW–410xv is a cost effective, easy to use and easy to configure IP communications solution for any business. The GXW–410xv supports popular voice codecs and is designed for full SIP compatibility and interoperability with 3rd party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology, integrate a traditional phone system into a VoIP network, and efficiently manage communication costs. This manual will help you learn how to operate and manage your GXW FXOAnalog IP Gateway and make the best use of its many upgraded features including simple and quick installation, multi-party conferencing, and direct IP-IP Calling. This IP Analog Gateway is very easy to manage and scalable, specifically designed to be an easy to use and affordable VoIP solution for the small – medium business or enterprise. Enable the video surveillance port to give piece of mind while you are away from your business. Gateway GXW-410xv Overview

The GXW410x offers an easy to manage, feature rich IP communications solution for any small business or businesses with virtual and/or branch locations who want to leverage their broadband network and/or add new IP Technology to their current phone system. The Grandstream Enterprise Analog VoIP Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls and vice versa. There are two models - the GXW-4104v and GXW-4108v, which have either 4 and 8 FXO ports respectively. The installation is the same for either model. A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-410xv series. In this environment, the SIP server handles SIP registration and call control and the GXW-410xv processes media conversion between IP and PSTN calls. By design, the system supports the call progress tones and PSTN signaling standards for North America, Europe, Latin America, Asia, and various other countries/regions. Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.

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• This document is contains links to Grandstream GUI Ingerfaces. Please remember to download these examples http://www.grandstream.com/GUI/GUI_GXW-410xv for your reference.

• This document is subject to change without notice. The latest electronic version of this user manual

is available for download from the following location: http://www.grandstream.com/user_manuals/GXW-410xv.pdf

• Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,

for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.

PACKAGING Unpack and check all accessories. Equipment included in the package:

aptor

AFETY COMPLIANCES nt with various safety standards including FCC/CE. Its power adaptor is

r agreement with our reseller customer. End users should contact the company

ndstream Sales and Service

ONNECTING THE GXW-410XV ACK PANEL

TABLE 1: DEFINITIO

1) One GXW-410xv Unit 2) One universal power ad3) One Ethernet cable

SThe GXW-410xv is compliacompliant with UL standard. Warning: use only the power adapter included in the GXW-410xv package. Using an alternative power adapter may permanently damage the unit.

ARRANTY WGrandstream has a resellefrom whom you purchased the product for replacement, repair or refund. f you purchased the product directly from Grandstream, contact your GraI

Representative for a RMA (Return Materials Authorization) number. Grandstream reserves the right to remedy warranty policy without prior notification. CFIGURE 1: DIAGRAM OF GXW-410XV B

NS OF THE GXW CONNECTORS

LAN/WAN RJ-45 Ethernet Ports VIDEO IN Jack

FXO PortsPower Supply

On/Off Switch

GXW-410xv

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LAN (or PC) Connect your PC to the LAN to find IP address from your Router/DHCP Server. The GXW-410xv acts as a switch. Connect to the internal LAN network or routWAN (or LAN) er.

VIDEO IN Connection for Analog based Video Surveillance Camera (RCA) RESET Factory Reset button. Press for 7 seconds to reset factory default settings. POWER IN Power adapter connection OFF/ON Off/On switch FXO1 - FXO8 FXO ports to be connected to physical PSTN lines from a traditional PSTN

PBX or PSTN Central Office. F 2: DIGURE IAGRA EL

ABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL

M OF GXW-410XV DISPLAY PAN

FXO port

GXW- 410xv

Display

T

Power LED Indicates Power. Remains ON when Power is connected and unit is turned ON.

Ready LED Remains ON after boot-up. LAN LED Indicates LAN (or WAN) port activity PC LED Indicates PC (or LAN) port activity Video LED Remains solid green on boot-up. If Video IN terminal is connected,

indicates video activity. LEDs 1 - 8 Indicate status of the respective FXO Ports on the back panel

Busy d Green) - ON (SoliAvailable - OFF

NOTE: All LEDs display green when ON.

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Application Description

A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-410xv series. In this environment, the SIP server handles SIP registration and call control and the GXW-410xv processes media conversion between IP and PSTN calls. By design, the system supports the call progress tones and PSTN signaling standards for North America, Europe, Latin America, Asia, and various other countries/regions. FIGURE 3: FUNCTIONAL DIAGRAM OF IP-PBX & GXW-410XV

PPSSTTNN CClloouudd

Anywhere in the world

GXW-410xv

4 or 8 Ports FXO Lines

PSTN Analog

Grandstream IP Phones

IPPBX or SIP Server

IP/LAN IP/WAN

SIMPLE CONFIGURATION: MEDIA GATEWAY TO ACCESS PSTN NETWORKS GXW-410xv can be configured to work with any leading SIP server, for a pure media gateway to access PSTN networks. In such applications, the user only needs to configure GXW-410xvgateway Stage Dialing field and Sip Server field. For a simple set-up, users only need to configure a SIP server field for default SIP Profile 1. This field should be configured to point to the SIP server to be used with the GXW-410xv. For advanced applications, the user is required to choose at least one SIP server field from the SIP profiles and one stage dialing under system Channel configuration table. On SIP server sides, the SIP server must be configured to forward user PSTN calls to the GXW-410xv. Please be aware that by default, the system uses North American PSTN settings and TWO STAGE dialing to access PSTN networks for VOIP to PSTN calls and PSTN to VOIP calls. Two stage dialing means the end-user will hear dial-tone twice. First dial-tone is used to let users to input destination number in the same network of the calling networks. Second dial-tone is used to let users to input final destination number.

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The GXW-410xv also supports ONE STAGE dialing, which means users only need to input one final

e

XTENSIVE CONFIGURATION: MEDIA GATEWAY CONFIGURATION FOR MULTIPLE USERS on

FF-HOOK AUTO DIAL y does not support direct inward dialing (DID). The GXW-410xv implements an

E CONFIGURATIONS - IP PBX PEERS WITH GXW-410XV

1) Configure GXW with SIP Accounts in IP PBX, this will enable you to put GXW behind a

2) without SIP Accounts in IP PBX, this makes GXW function as a PEER gateway.

ee the Quick Install Guide

PSTN number when first dial-tone is heard for calls from VOIP to PSTN. This requires configuring both SIP server and GXW-410xv to one stage dialing (see last section of quick guide and user manual for onstage dialing). For one stage PSTN to VOIP calls, user needs to configure off-hook auto dial field (see last section on sample configuration EThe GXW-410xv can be configured to work with a variety of SIP server features and traditional PBXPSTN networks, with a different SIP server on each physical port. Each port may have its own voice setting, dialing settings, PSTN termination setting, and DTMF transmission settings. OThe FXO interface currentloff-hook auto dial feature for each physical port. Configure off-hook auto dial to forward PSTN incoming call to a specific SIP number, call center or hunt group. SAMPL

There are 2 methods to configure GXW to work with IP PBX:

NAT/Firewall. Configure GXW

S at http://www.grandstream.com/user_manuals/GXW41xx_QuickIG.pdf for

ote: In regions other than North American, the user is also required to configure call progress tones se

more information for quick installation. Nand PSTN line termination fields. Check with local PSTN service carriers on values service providers uon the lines. If service provider doesn’t provide these values and users don’t know what the correct values are, please use the default values. Contact product support for additional help in configuring GXW-410xv.

your

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Features GXW-410xv is a next generation IP voice and video gateway that features full interoperability with leading IP-PBXs, SoftSwitches and SIP platforms. The Gateway series offers superb voice and video quality, traditional telephony functionality, simple configuration, feature rich functionality and an additional video port that enables the gateway to act like a video surveillance gateway. SOFTWARE FEATURES OVERVIEW

• 4 and 8 FXO port media gateways • Video surveillance port • External power supply • Two RJ-45 ports (switched or routed) • TFTP and HTTP firmware upgrade support • Multiple SIP accounts, multiple SIP profiles (3 profiles per system) • Supports Audio Codecs: G711, G723, G729 and GSM • Supports Video Codecs: H.264 • G.168 – echo cancellation • Flexible DTMF transmission: In Audio, RFC2833, SIP Info or any combination of the 3 • Selectable, multiple LBR coders per channel • T.38 compliant

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TABLE 3: GXW-410XV SOFTWARE FEATURES

GXW– 410x FX0 Analog Gateway Series Telephone Interfaces GXW-4104: 4 ports; 3 SIP accounts & profiles GXW-4108: 8 ports; 3 SIP accounts & profiles FXO, RJ11 Network Interface Two (2) 10/100 Mbps, RJ45 LED Indicators Power, Video, and Line LEDs On/Off Switch Yes Voice over Packet Capabilities

G.168 compliant Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711

Voice Compression G.711, G.723, G.726, G.729A/B, GSM Video Surveillance Real-time H.264 base CIF resolution DHCP Server/Client Yes, NAT Router or Switched Mode Fax over IP T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711

for Fax Pass-through (pending) QoS Diffserve, TOS, 802.1 P/Q VLAN tagging IP Transport RTP/RTCP & PPPoE PSTN Signaling FXO Loop start DTMF Method Flexible DTMF transmission method,

User interface of In-audio, RFC2833, and SIP Info IP Signaling SIP (RFC 3261) Provisioning TFTP and HTTP

Round-robin port scheduling to ensure available lines to access PSTN networks

Media SRTP Control TLS/SIPS (pending) Management Syslog support,

HTTPS and telnet (pending), remote management using Web browser Short and long haul REN3: Up to150 ft on 24 AWG line Caller ID Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID Polarity Reversal / Wink Yes EMC GXW-410x: EN55022 Class B, CFR Part 15 Class B, EN55024;

GXW-4104: FCC, CE (in addition) Safety GXW-410x: EN60950-1 GXW-4108: UL60950-1 (in addition)

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HARDWARE SPECIFICATION TABLE 4: HARDWARE SPECIFICATION OF GXW-410XV

LAN interface 2xRJ45 10/100Mbps

LED 8 LEDs (GREEN)

Universal Switching Power Adaptor

Input: 100-240V AC, 50/60Hz, 0.5A Max Output: 12V DC, 1.25A UL certified

Dimension 225mm (L) x 172mm (W) x 42mm (H)

Weight 0.29 lbs (3.5 oz)

Temperature 32~104°F 0~40°C

Humidity 10% - 90% (non-condensing)

Compliance FCC, CE

CONFIGURATION GUIDE CONFIGURATION WITH WEB BROWSER The GXW-410xv has an embedded Web server that will respond to HTTP GET/POST requests. It also has embedded HTML pages that allow a user to configure the IP phone through any common web browser. Examples of GUI interfaces can be downloaded @ http://www.grandstream.com/GUI/GUI_GXW-410xv. ACCESSING THE WEB CONFIGURATION MENU

1. Connect the Power to the GXW-410xv unit. 2. Connect an Ethernet cable between the LAN port on GXW-410xv to your PC. 3. You will have to assign a dummy IP with the same subnet as the GXW IP Address, which is

192.168.0.160 by default. So, set an IP address like 192.168.0.x for your PC. 4. Launch web browser and type http://192.168.0.160 at address of web browser. This connects you

to the GXW-410xv web server. You may choose to use DHCP or PPPoE connection or another static IP address according to your local network environment. The Gateway Web Configuration Menu can be then accessed by the following URI: http://Gateway-IP-Address where the Gateway-IP-Address is the IP address of the Gateway. NOTE: To access the configuration page, type the GXW IP address into the browser, stripping out the leading “0” because the browser will parse in octet. e.g. if the IP address is: 192.168.001.014, please type in: 192.168.1.14.

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END USER CONFIGURATION

Once this HTTP request is entered and sent from a Web browser, the GXW-410xv will respond with a login screen. There are two default passwords for the login page:

User Level: Password: Webpages allowed: End User Level 123 Only Status and Basic Settings Administrator Level admin All pages can be browsed.

FIGURE 4: SCREEN-SHOT OF GXW-410XV LOG-IN SCREEN

Grandstream Device Configuration

Password

Login

All Rights Reserved Grandstream Networks, Inc. 2005-2006 After login, the next configuration page is the Basic Configuration page, explained in detail in Table 6: Web Log-in Definition. TABLE 6: WEB LOG-IN DEFINITIONS

Web Access Select HTTP or secure HTTPS protocol for Web Access

Web Port By default, HTTP uses port 80 and HTTPS uses port 443. This field is for customizable web port.

End User Password This contains the password to access the Web Configuration Menu. This field is case sensitive with a maximum length of 25 characters.

IP Address There are two modes to operate the GXW-410xv: DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The GXW-410xv acquires its IP address from the first DHCP server it discovers from the LAN it is connected. Using the PPPoE feature: set the PPPoE account settings. The GXW-410xv will establish a PPPoE session if any of the PPPoE fields is set. Static IP mode: configure the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields. These fields are set to zero by default.

Time Zone Controls how the date/time is displayed according to the specified time zone.

You may also access the Device Status page which provides details of the GXW product. The Device Status page terms are defined in Table 7: Status Page Definitions.

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TABLE 7: STATUS PAGE DEFINITIONS

Hardware Revision Hardware version number: Main Board, Interface Board

MAC Address The device ID in HEX format. This is a very important ID for ISP troubleshooting.

IP Address This field shows LAN IP address of GXW-410xv

Product Model This field contains the product model info.

Software Version Program: This is the main software release. Boot and Loader are not changed often.

System Up Time This field shows system up time since the last reboot.

Registered This field indicates whether the different Channels are registered to the SIP server(s).

FXO Line Connected This field will give the status of each physical FXO Line connected to the Gateway. It will update the status regularly. Yes - Connected and Idle Busy - Connected and Busy No - Not connected

PPPoE Link Up This field shows whether the PPPoE connection is running if connected to DSL modem.

Detected NAT Type This field shows what kind NAT the GXW-410xv is connected to via its LAN port. It is based on STUN protocol.

ADVANCED USER SETTINGS ADVANCED USER CONFIGURATION

The end-user needs to login to the advanced user configuration page the same way as for the basic configuration page. FIGURE 5: SCREENSHOT OF ADVANCED USER CONFIGURATION

Grandstream Device Configuration

Password

Login

All Rights Reserved Grandstream Networks, Inc. 2005-2006 Advanced User configuration includes the end user configuration and advanced configurations including: SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.

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TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONS

Admin Password

Administrator password. Only the administrator can configure the “Advanced Settings” page. Password field is purposely left blank for security reasons. The maximum password length is 25 characters.

g723 Rate G723 encoding rate (6.3kbps or 5.3kbps)

Layer 3 QoS This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff-Serv or MPLS. Default value is 48.

Layer 2 QoS This contains the value used for layer 2 VLAN tag. Default setting is blank.

Inter Digit Timeout

Default is 4 seconds.

Local RTP port This parameter defines the local RTP-RTCP port pair the GXW-410xv will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.

Use Random Port

This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple GXW-410xvs are behind the same NAT.

Keep-alive interval

This parameter specifies how often the GXW-410xv sends a blank UDP packet to the SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.

Use NAT IP NAT IP address used in SIP/SDP message. Default is blank.

STUN Server IP address or Domain name of the STUN server.

Firmware Upgrade & Provisioning

This radio button will enable GXW-410xv to download firmware or configuration file through either TFTP or HTTP.

Via TFTP Server

This is the IP address of the configured TFTP server. If selected and it is non-zero or not blank, the GXW410x will attempt to retrieve new configuration file or new code image from the specified TFTP server at boot time. It will make up to 5 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image will be verified and then saved into the Flash memory. Note: Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device. Depending on the network environment this process can take up to 15 or 20 minutes.

Via HTTP Server

The URL for the HTTP server used for firmware upgrade and configuration via HTTP. For example, ttp://provisioning.mycompany.com:6688/Grandstream/1.0.0.36 Here “:6688” is the specific TCP port that the HTTP server is listening at, it can be omitted if using default port 80. Note: If Auto Upgrade is set to No, GXW-410xv will only do HTTP download once at boot up.

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Automatic Upgrade

Choose Yes to enable automatic upgrade and provisioning. In “Check for new firmware every” field, enter the number of days to enable GXW-410xv to check the server for firmware upgrade or configuration in the defined period of days. When set to No, GXW-410xv will only do upgrade once at boot up. “Always check for New Firmware.” Check New Firmware only when F/W pre/suffix changes”

Syslog Server The IP address or URL of System log server. This feature is especially useful for ITSP (Internet Telephone Service Provider)

Syslog Level Select the ATA to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events:

1. product model/version on boot up (INFO level) 2. NAT related info (INFO level) 3. sent or received SIP message (DEBUG level) 4. SIP message summary (INFO level) 5. inbound and outbound calls (INFO level) 6. registration status change (INFO level) 7. negotiated codec (INFO level) 8. Ethernet link up (INFO level) 9. SLIC chip exception (WARNING and ERROR levels) 10. memory exception (ERROR level)

The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GS_LOG: [device MAC address][error code] error message Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link is up

NTP server URI or IP address of the NTP (Network Time Protocol) server, which will be used by the phone to synchronize the date and time.

Enable Video Surveillance

Work in progress.

RTSP Port By default it is 554.

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Configuring the FXO channels on the GXW – 410xv is an easy process. Follow the GUI interfaces. The Device Status page terms are defined in Table 9: FXO Lines Configuration Definitions. An example of the Channel Dialing Configuration is shown in Figure 6. Please note the default is always configured. The user has the option to change the default settings as described in the Table 9. TABLE 9: FXO LINES CONFIGURATION DEFINITIONS

Enable Current Disconnect

When set to Y, Current Disconnect is enabled. Certain PSTN Cos require this to be enabled, in order to realize correct disconnect for PSTN side. Default it Y.

AC Termination Impedance

Selects the impedance of the analog Line connected to the FXO port on the GXW-410xv.

Silence Timeout Terminate call after long silence detected. Default is 60 seconds, max 65536.

DTMF Digit Length

Default value is 100ms.

DTMF Digit Volume

Default value is -11dB.

DTMF Dial Pause Default value is 100ms.

Wait Dial-tone It is recommended to set this to Y in case Dialing stage is set to 1.

Dialing Stage Dialing stage can be set to 1 or 2. Note: When set to 1, the Server needs to be configured to allow forwarding and receipt of SIP messages from GXW IP address directly.

Off-hook Auto Dial

This parameter allows users to configure a User ID or extension number to be automatically dialed upon off-hook. Please note only the user part of the SIP address needs to be entered here. The GXW-410xv will automatically append the ‘@’ and host portion of the corresponding SIP address.

FIGURE 6: SCREEN-SHOT OF GXW-410XV CHANNEL DIALING

Channel Dialing 1. DTMF Digit Length(X10ms):

ch1-8:10; (1-200, default 10)

ch1-8:-11; 2. DTMF Digit Volume(dB): (-31-0, default -11) ch1-8:10; 3. DTMF Dial Pause(X10ms): (1-200, default 10) ch1-8:Y; 4. Wait Dial-Tone(Y/N): (default Yes) ch1-8:2; 5. Dialing Stage(1/2): (default 2) ch1-8:605; 6. Off-hook Auto Dial(VoIP):

Update

All Rights Reserved Grandstream Networks, Inc. 2005-2006

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TABLE 10: CHANNELS PAGE DEFINITIONS

SIP User ID User account information, provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number.

Authentication ID SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or different from SIP User ID.

Authentication Password SIP service subscriber’s account password for GXW-410xv to register to (SIP) servers of ITSP.

Profile ID Select the corresponding Profile ID (1/2/3)

Call Progress Tones Using these settings, user can configure tone frequencies according to user preference. By default, the tones are set to North American frequencies. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds. ON is the period of ringing (ON time in ms) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. • “Dial tone” • “Ringback tone” • “Busy/Re order tone” • “Confirmation tone”

Channel Voice Settings Channel voice settings mentioned below.

Tx to PSTN Audio Gain (dB)

Allows user to set a value in dB for transmission to PSTN Audio Gain.

Rx from PSTN Audio Gain (dB)

Allows user to set a value in dB for receive from PSTN Audio Gain.

Silence Suppression This controls the silence suppression/VAD feature of G723 and G729. If set to “Yes”, when a silence is detected, small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to “No”, this feature is disabled.

Echo Cancellation When set to Y, Echo cancellation is enabled.

Channel specific Setting Channel specific settings mentioned below.

DTMF Method This parameter specifies the mechanism to transmit DTMF digits. There are7 modes supported: in audio which means DTMF is combined in audio signal (not very reliable with low bit-rate codec), via RTP (RFC2833), or via SIP INFO. Multiple DTMF transmission schemas can be selected.

1 – in-audio 2 – RFC2833 3 – in-audio and RFC2833 4 – SIP Info 5 – in-audio and RFC2833 6 – SIP Info and RFC2833 7 – in-audio, RFC2833, and SIP Info

No Key Entry Timeout Default is 4 seconds.

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TABLE 11: GRANDSTREAM CONFIGURATION PAGE DEFINITIONS

Activate Profile When set to Yes the SIP Profile is activated.

Profile Name A name to identify a Profile.

SIP Server SIP Server’s IP address or Domain name provided by VoIP service provider.

Outbound Proxy IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border Controller. Used by GXW-410xv for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work and ONLY outbound proxy can correct the problem.

Use DNS SRV: Default is No. If set to Yes the client will use DNS SRV to look up server.

User ID is Phone Number

If the GXW-410xv has an assigned PSTN telephone number, this field should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP request.

SIP Registration This parameter controls whether the GXW-410xv needs to send REGISTER messages to the proxy server. The default setting is “Yes”.

Unregister on Reboot

Default is No. If set to yes, the SIP user’s registration information will be cleared on reboot.

Register Expiration

This parameter allows the user to specify the time frequency (in minutes) for the GXW-410xv to refresh its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days).

Local SIP port This parameter defines the local SIP port the GXW-410xv will listen and transmit. The default value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and Account 4 respectively.

NAT Traversal This parameter defines whether the GXW-410xv NAT traversal mechanism will be activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, then the GXW-410xv will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the GXW-410xv will attempt to detect if and what type of firewall/NAT it is sitting behind through communication with the specified STUN server. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the GXW-410xv will attempt to use its mapped public IP address and port in all of its SIP and SDP messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the GXW-410xv will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the “hole” on the NAT open.

Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

Early Dial Default is No. Use only if proxy supports 484 response.

Session Expiration

Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

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Min-SE The minimum session expiration (in seconds). The default value is 90 seconds.

Caller Request Timer

If selecting “Yes” the phone will use session timer when it makes outbound calls if remote party supports session timer.

Callee Request Timer

If selecting “Yes” the phone will use session timer when it receives inbound calls with session timer request.

Force Timer If selecting “Yes” the phone will use session timer even if the remote party does not support this feature. Selecting “No” will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

UAC Specify Refresher

As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher.

UAS Specify Refresher

As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher.

Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer.

Enable 100rel The use of the PRACK (Provisional Acknowledgment) method enables reliability to be offered to SIP provisional responses (1xx series). This is very important if PSTN inter-networking is to be supported. A user’s request to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signalling messages.

Send Anonymous

If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will be set to anonymous, essentially blocking the Caller ID from displaying.

Preferred Vocoder

The GXW-410xv supports up to 5 different Vocoder types including G.711 A-/U-law, GSM, G.723.1, G.729A/B. The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message. The first Vocoder in this list can be entered by choosing the appropriate option in “Choice 1”. Similarly, the last Vocoder in this list can be entered by choosing the appropriate option in “Choice 8”.

Special Feature Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Broadsoft, etc.

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SAVING THE CONFIGURATION CHANGES

Once a change is made, press the “Update” button in the Configuration Menu. The GXW-410xv will display the following screen to confirm that the changes have been saved. Reboot or power cycle the GXW-410xv after all the changes are made so that those changes can take effect. FIGURE 7: SCREEN-SHOT OF SAVE CONFIGURATION

REBOOTING FROM REMOTE

The administrator can remotely reboot the unit by pressing the “Reboot” button at the bottom of the configuration menu. The following screen will indicate that rebooting is underway. FIGURE 8: SCREEN-SHOT OF REBOOTING

Grandstream Device Configuration

The device is rebooting now... You may re-login by clicking on the link below in 30 seconds.

Click to re-login

All Rights Reserved Grandstream Networks, Inc. 2005

The user can re-login to the unit after waiting for about 30 seconds.

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VIDEO SURVEILLANCE The GXW-410xv can be used with an Analog Surveillance Camera to perform video surveillance function. This application should be used in a LAN environment or when both sides have public IP address. VIDEO SURVEILLANCE PROCEDURES

Gateway side: 1. In the ADVANCED SETTING page, find the following field and change from default setting

NO to YES, reboot the device.

2. Connect an analog based surveillance camera to the VIDEOIN connection at the back panel

of the unit.

PC side (Monitor Device):

1. Download VLC from http://www.videolan.org/vlc/. This is the only player so far that works and support RFC 3984.

2. Launch VLC. 3. Go to Preferences->Input/Codecs->Demuxers->H264, check “Advanced options” in the

bottom. The option “Frames per Second” will show. Change that value to 5 and then save. 4. Go to Preferences->Input/Codecs->Access modules->Real RTSP, check “Advanced options”

in the bottom. The option “Caching value (ms) will show. Change that value to 1000 and then save. You may change it to a smaller value to reduce the delay.

5. If the viewer is under NAT, go to Preferences->Demuxers->Access modules->RTP/RTSP, check “Advanced options” in the bottom. The option “Use RTP over RTSP (TCP)” will show. Check that option box. (Grandstream does NOT recommend this network environment)

6. Close the Preferences window and go to File->Open Network Stream: a) Select RTSP as the protocol b) Enter the URL in the format of rtsp://admin:

ADMIN_PASSWORD@DEVICE_IP_ADDRESS:DEVICE_RTSP_PORT. Change the blue text according to your configuration:

• ADMIN_PASSWORD is the device’s web configuration password for admin. • DEVICE_IP_ADDRESS is the device IP. • DEVICE_RTSP_PORT is the RTSP port setting of the device.

If the port uses default value 554, the port portion can be omitted from the URL c) Click OK to start the video.

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FIGURE 9: SCREEN-SHOT OF VIDEO SURVEILLANCE*

* PC client side running VLC as monitoring station

FIRMWARE UPGRADE Our latest official release can be downloaded from: http://www.grandstream.com/y-firmware.htm. Firmware (or software) upgrades can be done either via TFTP or HTTP. The corresponding configuration settings are on the configuration page. End users should NOT touch the configuration settings that are useful for ITSPs. To upgrade your unit firmware, follow these steps:

1. Under Advanced Settings webpage, enter your TFTP or HTTP Server IP address (or FQDN) next to the “Firmware Upgrade: Upgrade Server” field.

2. Select via TFTP or HTTP accordingly.

3. If you plan to use Automatic Upgrade, set it to “Yes”, otherwise No (this will make it check for upgrade every time you reboot).

UPGRADE THROUGH HTTP To upgrade firmware via HTTP, the field “Firmware Upgrade and Provisioning: Upgrade Via” needs to be set to HTTP. The “Firmware Server Path” should be set to where the firmware files are located. For example, the user can use the following URL in the Firmware Server Path: firmware.mycompany.com: 6688/Grandstream/1.0.0.29 where firmware.mycompany.com is the FQDN of the HTTP server. It can also be in IP address format. “:6688” is the TCP port the HTTP server listening to, default http server listens to port 80. “/Grandstream/1.0.0.29” is the RELATIVE directory to the root dir on HTTP web server.

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UPGRADE THROUGH TFTP To upgrade firmware via TFTP, set the field “Firmware Upgrade and Provisioning: Upgrade Via” to TFTP. The TFTP server can be configured in either IP address format or FQDN. To configure the TFTP server via the Web configuration interface, follow these five steps:

1. Open your browser to input the IP address of the GXW-410xv. 2. Enter the admin password to enter the configuration screen. 3. Enter the TFTP server address or URL in the “Firmware Server Path” field near the

bottom of the configuration screen. 4. Once the “Firmware Server Path” is set, update the change by clicking the “Update”

button. 5. Reboot or power cycle the unit.

If the configured updating server is found and a new code image is available, the GXW-410xv will retrieve the new image files by downloading them into the GXW-410xv’s SRAM. During this stage, the GXW-410xv’s LED will blink until the checking/downloading process is completed. Upon verification of checksum, the new code image will be saved into the Flash. If TFTP fails for any reason (e.g., TFTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the GXW-410xv will stop the TFTP process and simply boot using the existing code image in the flash. Firmware upgrading may take as long as 20 minutes over the Internet, or just 20+ seconds if it is performed on a LAN. Grandstream recommends conducting firmware upgrades in a controlled LAN environment if possible. NO LOCAL TFTP SERVER

For users who do not have a local TFTP server, Grandstream provides a NAT-friendly TFTP server on the public Internet for users to download the latest firmware upgrade automatically. Please check the Services section of Grandstream’s Web site to obtain this TFTP server IP address. Alternatively, user can download and install a free TFTP or HTTP server in his LAN for a firmware upgrade. A free Windows version TFTP server can be downloaded from: http://support.solarwinds.net/updates/New-customerFree.cfm.

Directions For Downloading TFTP Server:

1. Unzip the file and put all of the files under the root directory of the TFTP server. 2. Put the PC running the TFTP server and the GXW–410x in the same LAN segment. 3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive

Only" to "Transmit Only" for the firmware upgrade. 4. Start the TFTP server, in the phone’s web configuration page. 5. Configure the Firmware Server Path with the IP address of the PC. 6. Update the change and reboot the unit.

You can also download the free HTTP server from http://httpd.apache.org/ or just use Microsoft IIS web.

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RESTORE FACTORY DEFAULT SETTING WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone. Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider. FACTORY RESET The ONLY way to restore default factory settings is as follows:

1. Unplug the Ethernet cable.

2. Locate a needle sized hole on the back panel of the gateway unit next to the Power connection.

3. Enter a needle like object in this hole and keep it pressed for about 7 seconds.

4. You will see the LAN port LEDs (green and orange) go off and on simultaneously; this indicates the reset went through.

5. All settings have been erased and the gateway is back to factory settings.

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Examples of GXW-410xv Configurations

APPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER

Scenario: A business with a traditional phone system (with or without broadband access) and an IP PBX or SIP Servers connecting to an Internet Telephone Service Provider (ITSP).

PPSSTTNN CClloouudd

Anywhere in the world

GXW-410xv

4 or 8 Ports FXO Lines

PSTN Analog

Grandstream IP Phones

IPPBX or SIP Server

IP/LAN IP/WAN

APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIO

Scenario: a small business with traditional analog PBX lines and broadband access who want to extend their traditional PBX to virtually anywhere in the world, using the internet. (Any SIP End point, such as Grandstream BugeTone, HandyTone, GXP-2000 or GXV-3000 are needed in this scenario)

Company A - Boston, MA 6 employees

Any branch, anywhere

IInntteerrnneett CClloouudd

IPPBX or SIP Server

Grandstream IP Phones

PPSSTTNN CClloouudd

IP/LAN

GXW-410xv

GXW-410xv

IP/LAN

FXS | IPPBX | SIP Platform

Any SIP endpoint

FXO Lines Traditional PBX

FXO Lines Optional

Anywhere in the world

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APPLICATION THREE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER AND VIDEO SURVEILLANCE

or

PPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATION

configuration,

Scenario: The GXW-410xv offers an additional video surveillance port which can be configured fsurveillance. It is the only small business analog gateway that offers this security feature.

A

Scenario Four: The GXW-410xv offers an IP to IP pure IP Communications Systemwhere all locations use IP phones.

Branch A – Boston, MA 6 employees

Branch B – Denver, CO 4 employees

IInntteerrnneett CClloouudd

IPPBX or SIP Server

Grandstream IP

PPSSTTNN CClloouudd

IP/LAN

GXW 410x

IPPBX or SIP Server

IP/LAN

Grandstream IP

Anywhere in the world

GXW 410x

Branch A - Boston, MA 6 employees

Branch B – Denver, CO 4 employees

IInntteerrnneett CClloouudd

IPPBX or SIP Server

Grandstream IP Phones

PPSSTTNN CClloouudd

IP/LAN

GXW 410x

IPPBX or SIP Server

IP/LAN

GXW 410x

Grandstream IP Phones

Anywhere in the world

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GLOSSARY OF TERMS ADSL Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800 kbps upstream, depending on line distance. AGC Automatic Gain Control is an electronic system found in many types of devices. Its purpose is to control the gain of a system in order to maintain some measure of performance over a changing range of real world conditions. ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The protocol operates below the network layer as a part of the interface between the OSI network and OSI link layer. It is used when IPv4 is used over Ethernet ATA Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP, like Grandstream HT series products. CODEC Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A) converter for translating the signals from the outside world to digital, and back again. CNG Comfort Noise Generator, generate artificial background noise used in radio and wireless communications to fill the silent time in a transmission resulting from voice activity detection.

DATAGRAM A data packet carrying its own address information so it can be independently routed from

ons of a signal in order to reduce the amount of information to be encoded or ompressed. Lossy compression algorithms ordinarily decimate while sub-sampling.

andard Institute from 1988, governing pan-European digital mobile telephony. ECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public

nd twelve full-duplex accesses per carrier, for a total f 120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all 12

ted from the RFP within a multi-frame (16 frames). Voice signals are igitally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation.

its source to the destination computer DECIMATE To discard portic DECT Digital Enhanced Cordless Telecommunications: A standard developed by the European Telecommunication StDswitched telephone network, Closed User Groups (CUGs), Local Area Networks, and wireless local loop. The DECT Common Interface radio standard is a multi-carrier time division multiple access, time division duplex (MC-TDMA-TDD) radio transmission technique using ten radio frequency channels from 1880 to 1930 MHz, each divided into 24 time slots of 10ms, aopossible accesses (time slots) simultaneously by using different frequencies or using only one frequency. All signaling information is transmitd DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain names into IP addresses

going rough an attendant or auto-attendant.

lized CPU used for digital signal processing. Grandstream roducts all have DSP chips built inside.

nals for dialing sing in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals

DID Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without th DSP Digital Signal Processor. A speciap DTMF Dual Tone Multi Frequency. The standard tone-pairs used on telephone termiusupport only 12 of them (0-9, * and #).

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FQDN Fully Qualified Domain Name. A FQDN consists of a host and domain name, including top-levdomain. For example,

el www.grandstream.com is a fully qualified domain name. www is the host,

Grandstream is the second-level domain, and and.com is the top level domain. FXO Foreign eXchange Office. An FXO device can be an analog phone, answering machine, fax, or anything that handles a call from the telephone company like AT&T. They should also operate the sway when connected to an FXS interface.

• An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions have their own standards.

ame

• FXO is complimentary to FXS (and the PSTN).

tation. An FXS device has hardware to generate the ring signal to the FXO xtension (usually an analog phone).

TS+PSTN for the phone. • The FXS Interface connects to FXO devices (by an FXO interface, of course).

rotocol (DHCP) is an Internet protocol for automating the , to vide

FXS Foreign eXchange Se

• An FXS device will allow any FXO device to operate as if it were connected to the phone company. This makes your PBX the PO

DHCP The Dynamic Host Configuration Pconfiguration of computers that use TCP/IP. DHCP can be used to automatically assign IP addressesdeliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to proother configuration information such as the addresses for printer, time and news servers. ECHO CANCELLATION Echo Cancellation is used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality on a telephone call. In addition to improving quality, this process improves bandwidth savings achieved through silence suppression by preventing echo from traveling across a network. There are two types of echo of relevance in telephony: acoustic echo and hybrid echo. Speech compression techniques and digital processing delay often contribute to echo generation in telephone networks. H.323 A suite of standards for multimedia conferences on traditional packet-switched networks.

yper Text Transfer Pr HTTP H otocol; the World Wide Web protocol that performs the request and retrieve function

IP-P IP T p general term for

e technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and

acket-switched etworks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony

P

s.

R IVR is a software application that accepts a combination of voice telephone input and touch-tone d

erhaps other media.

s of a server IP Internet Protocol. A packet-based protocol for delivering data across networks.

BX IP-based Private Branch Exchange

ele hony (Internet Protocol telephony, also known as Voice over IP Telephony) A thother forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet or other pnare often used to mean the same; however, they are not 100 per cent interchangeable, since Internet isonly a subcase of packet-switched networks. For users who have free or fixed-price Internet access, ITelephony software essentially provides free telephone calls anywhere in the world. However, the challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone system IVkeypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail anp

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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-t. The maximum for

thernet is 1500 byte.

AT Network Address Translation

bit bytes), that can be sent in a packet- or frame-based network such as the InterneE N NTP Network Time Protocol, a protocol to exchange and synchronize time over networks The port used is UDP 123 Grandstream products using NTP to get time from Internet OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.OBP/SBCs are put into the signaling and media path between calling and called party. The OBP/SBC acts as if it was the called VoIP phone and places a second call to the called party. The effect of this behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private OBP/SBCs are used along with

firewalls to enable VoIP calls to and from a protected enterprise nePublic VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks with

twork.

internet connections using NAT. PPPoE Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in Ethernet frames. It is used mainly with cable modem and DSL services. PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or

port Control Protocol, defined in RFC 3550

called POT (Plain Old Telephone), or circuit switched network. RTCP Real-time Trans , a sister protocol of the Real-time Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does not

riodically to transmit control packets to participants in a streaming ultimedia session. The primary function of RTCP is to provide feedback on the quality of service being

TP Real-time Transport Protocol defines a standardized packet format for delivering audio and video

transport any data itself. It is used pemprovided by RTP. Rover the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 SDP Session Description Protocol is a format for describing streaming media initialization parameterhas been published by the

s. It IETF as RFC 2327.

SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261). SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice

ansmission and uses fewer resources and is considerably less complex than H.323. All Grandstream

TUN Simple Traversal of UDP over NATs is a network protocol

trproducts are SIP based S allowing clients behind NAT (or multiple

et up UDP communication between two osts that are both behind NAT routers. The protocol is defined in RFC 3489

NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by the NAT with a particular local port. This information is used to sh . STUN will usually work well with non-symmetric NAT routers. TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCPapplications on networked hosts can create connections to one another, over which they can exchange data or

,

packets. The protocol guarantees reliable and in-order delivery of sender to receiver data. TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very basic form of FTP; It uses UDP (port 69) as its transport protocol.

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UDP User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages known as datagrams to one another.

DP does not provide the reliability and ordering guarantees that TCPU does; datagrams may arrive out of

ction or Voice Activity Detector is an algorithm used in speech processing

order or go missing without notice. However, as a result, UDP is faster and more efficient for many lightweight or time-sensitive purposes. VAD Voice Activity Dete

herein, the presence or absence of human speech is detected from the audio samples. w VLAN A virtual LAN, known as a VLAN, is a logically-independent network. Several VLANs can co-exist on a single physical switch. It is usually refer to the IEEE 802.1Q tagging protocol. VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of ignaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc. s

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